/*
 *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_GSM_INCLUDE_AUDIO_ENCODER_GSM_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_GSM_INCLUDE_AUDIO_ENCODER_GSM_H_

#include <vector>

#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h"
#include "webrtc/modules/audio_coding/codecs/gsmfr/include/gsmfr_interface.h"

namespace webrtc {

class AudioEncoderGSMFR : public AudioEncoder {
 public:
  struct Config {
   public:
    bool IsOk() const;

    int frame_size_ms;
    int num_channels;
    int payload_type;

   public:
	   Config()
        : frame_size_ms(10), num_channels(1), payload_type(3) {}
  };

  AudioEncoderGSMFR(const Config& config);

  ~AudioEncoderGSMFR() override;

  int SampleRateHz() const override;
  int NumChannels() const override;
  size_t MaxEncodedBytes() const override;
  int Num10MsFramesInNextPacket() const override;
  int Max10MsFramesInAPacket() const override;
  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                             const int16_t* audio,
                             size_t max_encoded_bytes,
                             uint8_t* encoded) override;

 private:
  const int sample_rate_hz_;
  const int num_channels_;
  const int payload_type_;
  const int num_10ms_frames_per_packet_;
  const size_t full_frame_samples_;
  std::vector<int16_t> speech_buffer_;
  uint32_t first_timestamp_in_buffer_;
  static const int kSampleRateHz = 8000;
  GSMFR_encinst_t_ *encoder_;
};

struct CodecInst;

class AudioEncoderMutableGSMFR
    : public AudioEncoderMutableImpl<AudioEncoderGSMFR> {
 public:
  explicit AudioEncoderMutableGSMFR(const CodecInst& codec_inst);
};

}  // namespace webrtc
#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_
